matrix telecom solutions: setu vfxth - fixed voip to fxo-fxs gateways
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Warm Welcome
Introduction
Configurations up to 32 FXO/FXS ports
Seamless connectivity between VoIP and PSTN network
Gateway for interfacing with existing IP-PBX or TDM PBX
Configurations up to 32 FXO/FXS ports
Seamless connectivity between VoIP and PSTN network
Gateway for interfacing with existing IP-PBX or TDM PBX
SETU VFXTHMulti-port VoIP-FXO-FXS Gateway
Interfaces
WAN Port
FXO/FXS Ports DC Input JackUSB Dongle
Multi-port VoIP-FXO-FXS Gateway
Multi-port VoIP-FXO-FXS GatewayVariants
Target Customers
Applications
SETU VFXTH
FXO/FXS
TDM PBX
PSTNFXS
1 - 32
IP
FXS
SIP Proxy
VoIP
VoIP Gateway for TDM PBX
IP
IP PBXSETU VFXTH
PSTNFXO
PSTN Gateway for IP-PBX
SETU VFXTH
FXS
TDM PBX
IP
PSTN
SETU VFXTH
PSTN
FXO
FXS
FXO
FXS
FXO
FXS
FXO
TDM PBX
FXSFXS FXSFXS
1-32 1-32
Peer-to-peer Application
Location 2
Location 3
Location 1
IP
PSTN
PSTN
FXO
FXS
FXS 32FXS 1
FXS
FXO
TDM PBX
SETU VFXTH
SETU VFXTH
IP PBX
IP Phone IP Phone
Location 4
FXS 1 FXS 32SETU VFXTH
Multi-Site Connectivity
Key Features
Can be programmed to allow or deny numbers from being dialed
Avoid misuse and restrict unproductive calls
Very useful feature to control cost
Can be programmed separately for VoIP, FXO and FXS ports
24 Number lists can be programmed
64 Entries per list are supported
Allowed and Denied List
Translates full number or part of a dialed number to match with numbering plan of
the destination network
Automatic Number Translation (ANT) is supported on VoIP, FXO and FXS ports User dials 001-xxxx to reach a number in the USA, but as the ITSP understands 1-xxxx so
ANT replaces the number 001-xxxx with 1-xxxx to let the ITSP understand the dialed
number string
Automatic Number Translation
Call details can be generated using various filters
Calls originated from FXO, FXS and SIP ports
Calls terminated on FXO, FXS and SIP ports
Calls made between with date, day and time
Calls with and without PIN authentication number
Called and calling party numbers matching with numbers list
Call duration with hour, minutes and seconds
Call details of 2000 calls can be stored in the buffer
Call details can be downloaded to a computer
Call Detail Record
Display calling party’s name and number on FXS port
Supports CLI on FXO and SIP ports
CLIP on call transfer
Supported CLI
DTMF
V.23 FSK
Bellcore FSK
Caller Line Identification
Different countries have different tones to indicate the process of call
activities such as Dial Tone, Ring Back Tone, Error Tone, Busy Tone etc.
Specific cadence can be programmed to match the tone used in each country
Call Progress Tones can be programmed country wise or can be customised as
per the user requirement
Call Progress Tones and Rings
Built-in feature
3-Party conference
Applicable if the call is originated from and terminated on FXS port
External and internal parties are possible
Conference
Daylight saving is the procedure of setting clocks ahead at a particular time of year
to make way for additional hour of daylight in the evening
Real Time Clock (RTC) moves backward or forward automatically in tune with the
daylight saving requirement of the country
DST can be forwarded or set backward according to day-month wise or date-month
wise as per the requirement
Day Light Saving
It is applicable on FXS ports
Do Not Disturb (DND) feature enables user to have privacy of not receiving the calls
for particular time period
Outgoing calls can be made when Do Not Disturb (DND) is enabled
Do Not Disturb
An industry standard for web authentication
Used to authenticate a caller in peer-to-peer network
Allows a group of devices to make and receive calls among them
Authentication password is to be pre configured in all the devices
Program such 500 entries in digest authentication table
Digest Authentication
Allows the caller to contact local emergency services like Police
Fire Station
Hospital
Maximum 5 emergency numbers can be programmed
Dialed out using pre-assigned port
Emergency Number Dialing
Send and receive FAX using IP and PSTN network
VoIP to PSTN FAX is possible if supported by ITSP
Send and receive Fax over IP using
T.38 (RTP)
T.38 (UDPTL)
Pass-Through
FAX over IP
Get connected to a pre-defined destination on picking up the receiver
Hotline is supported on FXS ports
Each FXS port can have different hotline number
Provision to impose delay before hotline activation
Delay timer can be minimum 1 and maximum 9 seconds
Hotline
Different service providers have different tariffs for different regions and countries
System automatically select the most economical route to place the call
Route selected depends on the destination number dialed
Ensure each call at least possible cost
Least Cost Routing
Allows making and receiving calls over IP network without the help of a proxy server
(Internet Telephony Service Provider)
Program destination number and destination address in the peer-to-peer table
Number can be of minimum 1 and maximum 24 digits
Useful for inter-branch or intra-office communication
Require internet connection with fixed IP Addresses
Peer-to-peer Calling
PIN Authentication is to authenticate a caller to prove identity to proceed the call
from one network to another
Supported on FXO and VoIP ports
Support 500 PIN authentication entries
Very useful feature to avoid misuse of the services
PIN Authentication
Date and time is very important parameter for some features like call detail record,
daylight saving time etc.
Real Time Clock uses the Simple Network Time Protocol (SNTP) to get time from the
time server
Flexibility to choose one from three pre-configured free reliable time servers
Flexibility to program time server address of their preference
Real Time Clock
System Log protocol is used for sending debug messages on IP network
It is a Client/Server protocol
Uses UDP as transport protocol for debugging process
Logging has several benefits
Easier and faster troubleshooting
Security enhancement
Better system administration
System Log
SETU VFXTH offers network connectivity like VoIP, FXO and FXS ports
Calls can be originated and terminated on any port type
Calls originated are routed to the destination as per programmed routing
mechanism
More than one destination port can be programmed for each source port
Three types of routing groups are supported
Universal Call Routing
Allowed and Denied Numbers List
Automatic Number Translation
Auto Provisioning for Mass Deployments
Call Detail Records
Call Progress Tone and Rings
CLI Based Call Routing
Dynamic DNS
Digest Authentication
Do Not Disturb
Emergency Number Dialing
Hotline
PCAP Trace
PIN Authentication
Message Wait Indication
SNMP Monitoring
SRTP/TLS over SIP
System Log Client
VLAN Tagging
Web based Programming
PBX Functionality Call Wait Call Hold Call Transfer Call Forward Call Pickup Conference
Feature List
Maximum Number of VoIP Channels 32 Channels
Maximum Number of FXO Ports (RJ11) 32 Ports
Maximum Number of FXS Ports (RJ11) 32 Ports
WAN Port (Ethernet Port) 1 Ethernet Port
Power Supply External Adaptor 24V DC/2.5A
Power Consumption 60 Watt (Typical)
LED Indications1 GREEN colour LED for Power, 1 Dual Colour LED for Status, 32 Single Colour LEDs for each Port
Dimensions (W x H x D) 40.7 X 5.1 X 17.2 Cm (16.0” X 2.0” X 6.8”)
Installation Mounting Table-Top, Wall and Rack Mount
Specifications
Matrix VoIP Product Range
PRODUCT DESCRIPTION
SETU ATA VoIP Adaptor with GSM, FXO and FXS Ports
SETU VGFX Multi-port SIP based VoIP to GSM/3G-FXO-FXS Gateway
SETU VGB Multi-port SIP based VoIP to GSM/3G-BRI Gateway
SETU VTEP SIP based VoIP to T1/E1 PRI Gateway
SETU VFX Low-Density Multi-port SIP based VoIP to FXO-FXS Gateway
SIMADO GFX Multi-port GSM to FXS Gateway
SIMADO GBR Multi-port GSM to BRI Gateway
Type of Presentation: Product Presentation
Number of Slides: 34
Revised On: 1st March 2013
Version-Release Number: V1R2
For Further Information Please Contact:
Email ID: Info.Telecom@MatrixComSec.com
Visit us at www.MatrixComSec.com
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